1. Field of the Invention
The present invention relates to apparatus and methods for multiband compression of sound input.
2. Description of the Prior Art
Multiband dynamic range compression is well known in the art of audio processing. Roughly speaking, the purpose of dynamic range compression is to make soft sounds louder without making loud sounds louder (or equivalently, to make loud sounds softer without making soft sounds softer). One well known use of dynamic range compression is in hearing aids, where it is desirable to boost low level sounds without making loud sounds even louder.
The purpose of multiband dynamic range compression is to allow compression to be controlled separately in different frequency bands. Thus, high frequency sounds, such as speech consonants, can be made louder while loud environmental noises--rumbles, traffic noise, cocktail party babble--can be attenuated.
The pending patent filed Oct. 10, 1995, Ser. No. 08/540,534 (herein incorporated by reference), entitled Digital Signal Processing Hearing Aid, inventors Melanson and Lindemann, gives an extended summary of multiband dynamic range compression techniques with many references to the prior art.
FIG. 1 (prior art) shows a block diagram of a conventional multiband compressor. The input signal from a microphone 104 or other audio source is divided into frequency bands using a filter bank 106 made up of a plurality of band pass filters, of which three are shown here: 108, 110, and 112. Most multiband compressors in analog hearing aids have two or three frequency bands.
A power estimator (122, 124, 126) estimates the power of each frequency band (114, 116, 118) at the output of each band pass filter. These power estimates are input to a plurality of gain calculation blocks (130, 132, 134) which calculate a gain (138, 140, 142 ) which will be applied to the frequency bands 114, 116, 118. In general, gains 138, 140, and 142 provide more gain for low power signals and less gain for high power signals. The gain is multiplied with the band pass signal and the gain scaled band pass signals 146, 148, 150 are summed by adder 154 to form the final output. This output will generally be provided to a speaker or receiver 158.
When dividing an audio signal into frequency bands, it is desirable to design the filter bank in such a way that, if equal gain is applied to every frequency channel, the sum of the frequency channels is equal to the original input signal to within a scalar gain factor. The frequency response of the sum of the frequency channels should be nearly constant. In practice we can tolerate phase distortion better than amplitude distortion so we will say that the magnitude frequency response of the sum of frequency channels should be nearly constant. Less than 1 dB of ripple is desirable.
FIG. 2 shows the magnitude frequency response of the band pass channels 201 and the magnitude frequency response of the sum of band pass channels 202 of a filter bank designed in the manner described above. In U.S. Pat. No. 5,500,902, Stockham Jr. et al. propose just such a filter bank as the basis of a multiband compressor. The band centers and bandwidths of the filter bank are spaced roughly according to the critical bands of the human ear. This is a quasi-logarithmic spacing--linear below 500 Hz and logarithmic above 500 Hz. It is suggested in U.S. Pat. No. 5,500,902 in column 5 lines 8-9 that the audio band pass filters should preferably have a band pass resolution of 1/3 octave or less. In other words, the band pass filters should be reasonably narrow as indicated in FIG. 2 so that the compression is controlled independently in each band with little interaction between bands.
FIG. 3 shows the magnitude frequency response of the sum of frequency channels 202 for the same filter bank as FIG. 2, but with higher resolution on the Y axis. We can see that the residual ripple is considerably less than 1 dB.
When a multiband compression system, based on such a filter bank, is presented with a broadband signal, such as white noise, it will adjust the gain similarly in each frequency channel. The gains may be weighted so that the wider bands at high frequency, which measure more power because of their increased width, produce gains equivalent to the narrow low frequency bands. The result is a smooth, flat output frequency response.
However, when such a filter bank is presented with a narrow band stimulus, such as a sinusoid slowly swept across frequency, the resulting output response is entirely different, as shown in FIG. 4. The sine wave is swept slowly enough so that the time constants of the compressor are not a factor. We see a pronounced 4.5 dB ripple in the output 401. Here the stimulus is a -20 dB sinusoid sweeping across frequency. The compression ratio in this example is 4 to 1 and the unity gain point of the compressor is 0 dB. Under these conditions, we would expect the compressor to generate 15 dB of gain so that the resulting output is a constant -5 dB. This is clearly not the case.
As we recall, the filter bank is designed to sum to a constant response. This means at the filter crossover frequencies, where the response of adjacent band pass filters is the same, the band pass response is -6 dB. Since the responses are the same at this point they will sum, giving a total of 0 dB which preserves the overall flat response. However, when a sinusoid is presented at a crossover frequency the power measurement is also -6 dB relative to the band center. The compressor in each band sees this -6 dB output and, since the compression ratio is 4 to 1, generates a gain of 4.5 dB which appears on the output as shown in FIG. 4. Note that the ripple would be smaller for a system having a lower compression ratio. For a compression ratio of 1.5, the ripple would be around 2 dB, which is still quite significant.
For narrow band signals which change frequencies this will generate an undesirable audible warble. This would certainly be the case for musical sounds--flutes, violins, etc. It would also be the case for high pitched speech sounds from women and children where the individual harmonics of voiced speech are relatively far apart and will appear as individual stimuli. As the formants of the voiced speech sweep across frequency they will become distorted by the narrow band ripple shown in FIG. 4.
In addition, audiologists often test the frequency response of hearing aids with pure tone sinusoids of different frequencies. The results of their tests will clearly be compromised given the response of FIG. 4.
For illustrative reasons, in FIG. 5 we have decreased the number of bands to three bands, 501, 502, and 503. This is considerably fewer bands than the FIG. 2 configuration, but the filter bands are conventionally overlapped, and the ripple or warble problem remains the same as in the FIG. 2 configuration. In FIG. 5, the filter transfer functions are plotted using different symbols for each filter. Thus, frequency band 501 is plotted with squares, frequency band 502 is plotted with triangles, and frequency band 503 is plotted with asterisks. The band transitions in the FIG. 5 configuration are relatively sharp and there is just enough overlap to guarantee that the sum of the magnitude frequency responses of the filters is constant, as shown by 504, which indicates the broadband frequency response of the configuration. However, as shown in FIG. 6, the slowly swept sine response 601 of the 4 to 1 compressor manifests a 4.5 dB ripple, just as was seen in FIG. 4.
This poor response to narrow band inputs is true for any compressor with relatively narrow transition bands (conventional overlap) between band pass filters. In particularly it is true for both digital and analog hearing aids with two or more frequency channels.
A need remains in the art for a multiband dynamic range compressor which is well behaved for narrow band and broad band signals.